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Unread 12-26-2009, 05:18 PM   #2
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Originally Posted by alfonso View Post
If I use the following configuration: MODEM>>LINKSYS WRT320N>>SIPURA SPA3102 then the bell rings but there is no voice coming through.
Are you speaking of receiving the SIP call directly, not through your Voxalot number?

The problem you mentioned is likely caused by your SPA3102 sending it's local IP address and port as RTP connection information (found inside SIP headers). When this happens the remote SIP device often does not send any voice packets towards you, so you hear nothing. Ordinarily, RTP connection information should be given as an external WAN IP address and port number. The following settings may fix your problem.

(under SIP tab)
SIP_T1: 1
Handle_VIA_received: yes
Handle_VIA_rport: yes
Insert_VIA_received: yes
Insert_VIA_rport: yes
Substitute_VIA_Addr: yes
Send_Resp_To_Src_Port: yes
STUN_Enable: yes
STUN_Test_Enable: yes
STUN_Server: (or any STUN server such as '' or '')
NAT_Keep_Alive_Intvl: 179 (if your phone does not always ring on incoming calls when using '179' try 119 or 59, use highest number that works)

(under Line_1 and Line_2 (or PSTN_Line) tabs)
NAT_Mapping_Enable: yes
NAT_Keep_Alive_Enable: yes
NAT_Keep_Alive_Msg: 0000
NAT_Keep_Alive_Dest: $PROXY
Register Expires: 3600


Optional settings, steps 1 - 4:
Sometimes its preferable to operate the ATA independently of a STUN server. The benefit is that phone service can continue no matter if the STUN server is working. See steps 1 to 4 below.

1. Forward the SIP ports and the RTP port range from the router to the ATA.
2. Set "STUN Enable:" no
3. Set "NAT Keep Alive Enable:" no
4. The ATA must learn it's public IP address, either through normal SIP registration, or place your public IP address in the 'EXT IP:' field).
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